1. Field of the Invention
The present invention is directed in general to television modulators. In one aspect, the present invention relates to a method and system for digitally filtering audio signal information in accordance with established standards for the broadcast of stereophonic cable and television signals in the United States and in other countries. In a further aspect, the present invention provides an integrated circuit system for filtering audio signals using allpass decomposition Cauer low pass filters to generate an aural composite signal in a digital BTSC encoder.
2. Related Art
In 1984, the Federal Communications Commission (FCC) adopted a standard for the audio portion of television signals called Multichannel Television Sound (MTS) which permitted television programs to be broadcast and received with bi-channel audio, e.g., stereophonic sound. Similar to the definition of stereo for FM radio broadcast, MTS defined a system for enhanced, stereo audio for television broadcast and reception. Also known as BTSC stereo encoding (after the Broadcast Television System Committee (BTSC) that defined it), the BTSC transmission methodology is built around the concept of companding, which means that certain aspects of the incoming signal are compressed during the encoding process. A complementary expansion of the signal is then applied during the decoding process.
The original monophonic television signals carried only a single channel of audio. Due to the configuration of the monophonic television signal and the need to maintain compatibility with existing television sets, the stereophonic information was necessarily located in a higher frequency region of the BTSC signal, making the stereophonic channel much noisier than the monophonic audio channel. This resulted in an inherently higher noise floor for the stereo signal than for the monophonic signal. The BTSC standard overcame this problem by defining an encoding system that provided additional signal processing for the stereophonic audio signal. Prior to broadcast of a BTSC signal by a television station, the audio portion of a television program is encoded in the manner prescribed by the BTSC standard, and upon reception of a BTSC signal, a receiver (e.g., a television set) then decodes the audio portion in a complementary manner. This complementary encoding and decoding ensures that the signal-to-noise ratio of the entire stereo audio signal is maintained at acceptable levels.
FIG. 1 is a block diagram of the front end portion of an analog BTSC encoding system 100, as defined by the BTSC standard. Encoder 100 receives left and right channel audio input signals (indicated in FIG. 1 as “L” and “R”, respectively) and generates a conditioned sum signal (“L+R”) and an encoded difference signal (“L−R”). It should be appreciated that, while the system of the prior art and that of the present invention is described as useful for encoding the left and right audio signals of a stereophonic signal that is subsequently transmitted as a television signal, the BTSC system also provides means to encode a separate audio signal called SAP (secondary audio program), e.g., audio information in a different language, which is separated and selected by the end receiver. Further, noise reduction components of the BTSC encoding system can be used for other purposes besides television broadcast, such as for improving audio recordings.
System 100 includes an input section 110, a sum channel processing section 120, and a difference channel processing section 130. Input section 110 receives the left and right channel audio input signals and generates a sum signal (indicated in FIG. 1 as “L+R”) and a difference signal (indicated in FIG. 1 as “L−R”). It is well known that for stereophonic signals, the sum signal L+R may be used by itself to provide monophonic audio reproduction and it is this signal that is decoded by existing monophonic audio television sets to reproduce sound. In stereophonic receivers, the sum and difference signals can be added to and subtracted from one another to recover the original two stereophonic signals (L) and (R). Input section 110 includes two signal adders 112, 114. Adder 112 sums the left and right channel audio input signals to generate the sum signal, and adder 114 subtracts the right channel audio input signal from the left channel audio input signal to generate the difference signal.
To accommodate transmission path conditions for television broadcasts, the difference signal is subjected to additional processing than that of the sum signal so that the dynamic range of the difference signal can be substantially preserved as compared to the sum signal. More particularly, the sum channel processing section 120 receives the sum signal and generates the conditioned sum signal. Section 120 includes a 75 μs preemphasis filter 122 and a bandlimiter 124. The sum signal is applied to the input of filter 122 which generates an output signal that is applied to the input of bandlimiter 124. The output signal generated by the latter is then the conditioned sum signal.
The difference channel processing section 130 receives the difference signal and generates the encoded difference signal. Section 130 includes a fixed preemphasis filter 132 (shown implemented as a cascade of two filters 132a and 132b), a variable gain amplifier 134 preferably in the form of a voltage-controlled amplifier, a variable preemphasis/deemphasis filter (referred to hereinafter as a “variable emphasis filter”) 136, an overmodulation protector and bandlimiter 138, a fixed gain amplifier 140, a bandpass filter 142, an RMS level detector 144, a fixed gain amplifier 146, a bandpass filter 148, an RMS level detector 150, and a reciprocal generator 152. The processing of the difference signal (“L-R”) by section 130 is substantially as described in the Background section of U.S. Pat. No. 5,796,842, which explains that the BTSC standard rigorously defines the desired operation of the 75 μs preemphasis filter 122, the fixed preemphasis filter 132, the variable emphasis filter 136, and the bandpass filters 142, 148, in terms of idealized analog filters. Specifically, the BTSC standard provides a transfer function for each of these components and the transfer functions are described in terms of mathematical representations of idealized analog filters. The BTSC standard further defines the gain settings, Gain A and Gain B, of amplifiers 140 and 146, respectively, and also defines the operation of amplifier 134, RMS level detectors 144, 150, and reciprocal generator 152. The BTSC standard also provides suggested guidelines for the operation of overmodulation protector and bandlimiter 138 and bandlimiter 124. Specifically, bandlimiter 124 and the bandlimiter portion of overmodulation protector and bandlimiter 138 are described as low pass filters with cutoff frequencies of 15 kHz, and the overmodulation protection portion of overmodulation protector and bandlimiter 138 is described as a threshold device that limits the amplitude of the encoded difference signal to 100% of full modulation where full modulation is the maximum permissible deviation level for modulating the audio subcarrier in a television signal.
To create the stereo signal, the BTSC standard also defines a composite stereophonic baseband signal (referred to hereinafter as the “composite signal”) that is used to generate the audio portion of a BTSC signal. The composite signal is generated using the conditioned sum signal (“L+R”), the encoded difference signal (“L−R”), and a tone signal, commonly referred to as the “pilot tone” or simply as the “pilot,” which is a sine wave at a frequency Fp, where Fp is equal to 15,734 Hz.
FIG. 2 is a graph of the spectrum of the composite signal. In FIG. 2, the spectral band 202 containing the content of the conditioned sum signal (or the “sum channel signal”) is indicated as “L+R.” The spectral sideband 204 containing the content of the frequency shifted encoded difference signal (or the “difference channel signal”) is each indicated as “L−R,” and the pilot tone 210 is indicated by the arrow at frequency Fp. In addition, the spectral sideband 206 containing the content of the frequency shifted encoded secondary audio program (or the “secondary audio channel”) is each indicated as “SAP,” and the spectral sideband 208 containing the content of the frequency shifted professional channel is each indicated as “Professional Channel.” As shown in FIG. 2, the encoded difference signal is used at 100% of full modulation, the conditioned sum signal is used at 50% of full modulation, and the pilot tone is used at 10% of full modulation.
The encoded “L+R” and “L−R” signals are transmitted to the receiver, such as a stereo television set, where a stereo decoder uses both the “L+R” and “L−R” signals in a matrix that decodes and restores the original L and R audio program. For purposes of transmitting a BTSC encoded signal, a third signal, called the pilot subcarrier signal 210, is inserted between the main-channel signal 202 (L+R) and the stereo signal 204 (L−R), as illustrated in FIG. 2. According to the BTSC standard, the pilot subcarrier shall be frequency locked to the horizontal scanning frequency of the transmitted video signal, and may be used to indicate the presence of multiple sound channels or to process these sound channels at the receiver. The composite signal is generated by multiplying the encoded difference signal by a waveform that oscillates at twice the pilot frequency according to the cosine function cos(4π Fpt), where t is time, to generate an amplitude modulated, double-sideband, suppressed carrier signal and by then adding to this signal the conditioned sum signal and the pilot tone.
In the past, BTSC stereo encoders and decoders were implemented using analog circuits. Through careful calibration to tables and equations described in the BTSC standard, the encoders and decoders could be matched sufficiently to provide acceptable performance. However, conventional analog BTSC encoders (such as described in U.S. Pat. No. 4,539,526) have been replaced by digital encoders because of the many benefits of digital technology. Prior attempts to implement the analog BTSC encoder 100 in digital form have failed to exactly match the performance of analog encoder 100. This difficulty arises from the fact that the BTSC standard defines all the critical components of idealized encoder 100 in terms of analog filter transfer functions, and prior digital encoders have not been able to provide digital filters that exactly match the requirements of the BTSC-specified analog filters. As a result, conventional digital BTSC encoders (such as those described in U.S. Pat. Nos. 5,796,842 and 6,118,879) have deviated from the theoretical ideal specified by the BTSC standard, and have attempted to compensate for this deviation by deliberately introducing a compensating phase or magnitude error in the encoding process.
An additional drawback with conventional digital encoders is the complexity and performance problems associated with digitally filtering the audio signals in a signal encoder. For example, direct mapping of analog filters into the digital domain can result in signal distortion from frequency warping in the filter response. Another problem occurs when there is inadequate attenuation in a frequency region of interest, such as with the audio low pass filters referenced in the BTSC standard for preventing crosstalk into other channels or into the pilot spectrum space. Conventional low order Cauer low pass solutions (such as suggested in “Multichannel Television Sound—BTSC System Recommended Practices,” EIA Television Systems Bulletin No. 5, Section 2.4.1 (July 1985)) require additional filtering (such as notch filters for the BTSC pilot frequency) or have used finite impulse response filters with insufficient cutoff. Other Cauer filter solutions have used infinite impulse response filters that suffer from limit cycle behavior as the filter order increases or that require processing and truncation of additional bits so that the least significant bits are ignored.
In addition to the complexity of the computational requirements for encoding the stereo signals, such as described above, the ever-increasing need for higher speed communications systems imposes additional performance requirements and resulting costs for BTSC encoding systems. In order to reduce costs, communications systems are increasingly implemented using Very Large Scale Integration (VLSI) techniques. The level of integration of communications systems is constantly increasing to take advantage of advances in integrated circuit manufacturing technology and the resulting cost reductions. This means that communications systems of higher and higher complexity are being implemented in a smaller and smaller number of integrated circuits. For reasons of cost and density of integration, the preferred technology is CMOS. To this end, digital signal processing (“DSP”) techniques generally allow higher levels of complexity and easier scaling to finer geometry technologies than analog techniques, as well as superior testability and manufacturability.
Conventionally known audio encoding systems, such as BTSC encoders, have not provided adequate digital filtering during audio signal encoding. Further, the nature of existing analog BTSC encoders has made them inconvenient to use with digital equipment such as digital playback devices. A digital BTSC encoder could accept the digital audio signals directly and could therefore be more easily integrated with other digital equipment. Therefore, there is a need for a better system that is capable of performing the above functions and overcoming these difficulties without increasing circuit area and operational power. Further limitations and disadvantages of conventional systems will become apparent to one of skill in the art after reviewing the remainder of the present application with reference to the drawings and detailed description which follow.